Public Switched Telephone Network (PSTN) telephony offers only one level of service based on the use of a single audio quality regime. For example, the PSTN uses 64 kilobits Per Second (BPS) Pulse Code Modulated (PCM) audio with a 300–3400 Hertz passband. PSTN telephony further offers only one control regime that deterministically either grants or denies service rather than degrading service when a failure or network congestion occurs.
Voice-Over-Internet Protocol (VoIP) systems are often built to mimic this service regime but are also inherently more flexible. Certain VoIP protocols and algorithms can adapt to varying service levels. Example adaptations include either denying service or degrading service when resources cannot be reserved, or reserved resources become unavailable. Different encoder/decoders (Codecs) are switched in and out if bandwidth becomes more or less scarce. Forward Error Correction (FEC) can be added or removed as packet error rates change. Packetization intervals can also be varied to either limit bandwidth utilization or limit packet transmission rate.
To date these adaptations have been statically provisioned by the network designer and are not user controllable. The problem is that these adaptations may not be optimized for the current telephone users or the current VoIP call. For example, a given quality of service may be more than adequate for one user. However, that same given quality of service may be unsatisfactory to a different user.
The present invention addresses this and other problems associated with the prior art.